Measurements sample rate and filter resolution

[Leo11]

Hello everyone.
I have a doubt about the process of measuring and creating filters, especially in light of the latest SW releases.
If I make measurements at 96kHz, for example, and export the filter with 65K taps, I will get about 1.5Hz resolution.
While with 44kHz it should be about 0.7Hz.
Now, if I resample externally the wav file of the filter generated with 96kHz measurements to 44kHz, does its resolution change o it's always as the original (1.5hz)?
I ask this because I have my 96khz filters resampled at 44khz, and I don't know if it is appropriate to repeat the whole filter measurement and creation procedure to get the best resolution.

 

[juicehifi]

If you use the new resampling in Audiolense, the resolution you see on the screen will be maintained. If you use the old approach (resample measurement first), the filter-length will be kept, but the resolution wil be changed.

If you resample externally, the resolution will be maintained, unless the resampler have some length limit embedded that prevents this.

 

[Leo11]

Okay. Thanks for the reply.

I take this opportunity to ask another question about the function to make the subs mono.
Typically the audio tracks have low frequencies already in mono below 100Hz, where the stereo content in fact would be useless for spatial perception.
When you have a setup with 2 subs or more, considering the high wavelength and neglecting the alterations of the room, the low frequency sound pressure of each sub is added.
So if we have 2 channels that individually emit an 80Hz tone at 90dB at LP, the result is about 93dB at LP.
Now, in the real world the two subs will have a different frequency response, so in the LP the sum will not provide linear response.
For example, if a channel has an 80Hz dip due to cancellation, so the LP pressure is 80dB instead of 90dB from the previous case, the sum will be about 90.4dB at 80Hz. But the rest of the frequencies (assuming they have a 90dB flat response for each channel) is always at 93dB.
In this case to perceive a full flat response at 93dB (always considering that the bass in audio tracks is typically mono) the other sub not affected of cancellation could (or should) emit 80Hz at 92.8dB to compensate (80 + 92.8 = 93).
And so on if there are multiple subs.
This is because the bass in fact cannot be localized so it is indifferent to where the pressure comes from.

The question is, Audiolense integrates this type of correction when I select the option to route bass in mono, or each channel is corrected according to the individual response? (therefore with the limitations due to cancellations)

 

[juicehifi]

Each sub is corrected as a single entity. But that only applies to when the subs act as a separate speaker. This is strictly limited to lfe.

When several subs are used to offload any main speaker, the sum of subs is corrected together with the main speaker. The fluctiations of a single sub will be significantly downplayed. However, if you have serious dips in one sub at a certain frequency it may influence the overall result.

You can also configure several subs in Audiolense as one multichannel sub, e.g where one or two subs share the upper half and one or two other shares the lower half. This may be a way to work around difficult cancellations.

Nevertheless, any close to perfect solution is one where the speaker placements and the acoustics are so good that each speaker as well as the total is easily correctable. It takes great acoustics to achieve great soun.

 

[Leo11]

Thanks for the reply.
I have been busy elsewhere lately and have no way of experimenting.
Anyway, one thing is not clear to me.
I have two speakers and two subs (equal).
If I wanted to flat the summed response of the two subs, each feeded with 0.5L+0.5R, but also individually corrected in time domain, how I have to configure they?
It is correct to set they as two independent unit ("Subwoofer") and check the option "Route bass to mono"?
Because the alternative that comes to mind is to create a single subwoofer unit made up of two drivers, totally overlapping in the range. With the TTD option for drivers they should be corrected individually in the time domain, but the response is corrected as a single unit (right?).
In this case however I can't understand what would happen in the crossover region with each main speaker...